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zijiehe b68d655f36 Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions 
I have chosen part of 2435603010 changes to compose this change.
According to the discussion we have made in previous change, this CL contains,
1. Source structure to represent a source of a DesktopCapturer.
2. GetSourceList / SelectSource / FocusOnSelectedSource functions in
DesktopCapturer.
3. ScreenCapturer and WindowCapturer forward corresponding functions to the new
DesktopCapturer APIs.

After this change, We can remove WindowCapturer & ScreenCapturer references from
Chromium, and use the new APIs.

BUG=webrtc:6513

Committed: https://crrev.com/9cb0b3b4ac916cdf52d97a63d923dfbe73f0541e
Review-Url: https://codereview.webrtc.org/2452263003
Cr-Original-Commit-Position: refs/heads/master@{#14830}
Cr-Commit-Position: refs/heads/master@{#14832}
2016-10-29 00:35:16 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Fixing flaky DtmfSenderTest by using fake clock.
2016-10-28 20:53:17 +00:00
audio
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
base
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
build
Restore symbol level for Android builds.
2016-10-27 07:00:20 +00:00
call
Revert of Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #2 id:20001 of https://codereview.webrtc.org/2455963004/ )
2016-10-28 21:06:36 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Reland of Move bitstream parser to more appropriate directory. (patchset #1 id:1 of https://codereview.webrtc.org/2430353004/ )
2016-10-26 09:48:24 +00:00
examples
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
libjingle
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
Revert of Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #2 id:20001 of https://codereview.webrtc.org/2455963004/ )
2016-10-28 21:06:36 +00:00
modules
Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions
2016-10-29 00:35:16 +00:00
p2p
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
pc
Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit.
2016-10-25 17:50:41 +00:00
sdk
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
stats
RTCStats equality operator added.
2016-10-25 11:31:27 +00:00
system_wrappers
Delete DataLog abstraction, which was almost unused.
2016-10-21 08:53:01 +00:00
test
Remove use of VoECodec in video/call tests.
2016-10-27 07:23:14 +00:00
tools
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
video
Revert of Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #2 id:20001 of https://codereview.webrtc.org/2455963004/ )
2016-10-28 21:06:36 +00:00
voice_engine
Original CL: https://codereview.webrtc.org/2433153003/, commit 8b8d3e4c30e8ea3846b58dfd36d1fd35a7799df4.
2016-10-28 14:55:39 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
call.h
Integrate FlexfecReceiveStream with Call.
2016-10-24 06:37:24 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common_types.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
config.h
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_frame.h
Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 07:44:17 +00:00
video_receive_stream.h
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +00:00
video_send_stream.h
Implement framesEncoded stat in video send ssrc stats.
2016-10-24 09:09:39 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +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.