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kthelgason ff046c74c5 Remove ALL usage of CriticalSectionWrapper. 
Finally we are able to remove this class entirely, along with the last
vestiges of it's use. I've also removed some legacy files that were only
used for windows XP support.

BUG=webrtc:7035

Review-Url: https://codereview.webrtc.org/2790533002
Cr-Commit-Position: refs/heads/master@{#17480}
2017-03-31 09:03:55 +00:00
..
api
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
audio
Add Windows, Mac, Android support to low bandwidth audio test
2017-03-30 11:01:30 +00:00
base
Reland: Use native (optimized) functions for byte order conversion. (patchset #1 id:1 of https://codereview.webrtc.org/2755103002/ )
2017-03-31 08:42:09 +00:00
call
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
common_audio
iSAC fix entropy coder: Recently added DCHECK could in fact trigger
2017-03-13 12:28:47 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Revert of Enable the bayesian bitrate estimator by default. (patchset #5 id:80001 of https://codereview.webrtc.org/2749803002/ )
2017-03-30 17:36:53 +00:00
logging
To accommodate some downstream WebRTC users we need to loosen
2017-03-30 08:24:20 +00:00
media
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
modules
Remove ALL usage of CriticalSectionWrapper.
2017-03-31 09:03:55 +00:00
ortc
Disable the ORTC integration tests on TSan.
2017-03-28 20:45:18 +00:00
p2p
Disable possibly flaky AudioCodingModuleTest.TestPacketLossStereo for Linux.
2017-03-28 10:00:07 +00:00
pc
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
sdk
Revert of Enable the bayesian bitrate estimator by default. (patchset #5 id:80001 of https://codereview.webrtc.org/2749803002/ )
2017-03-30 17:36:53 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Remove ALL usage of CriticalSectionWrapper.
2017-03-31 09:03:55 +00:00
test
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
tools
To accommodate some downstream WebRTC users we need to loosen
2017-03-30 08:24:20 +00:00
video
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
voice_engine
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
.gitignore
…
BUILD.gn
Add Windows, Mac, Android support to low bandwidth audio test
2017-03-30 11:01:30 +00:00
call.h
…
codereview.settings
…
common_types.cc
…
common_types.h
Delete unused constant kConfigParameterSize.
2017-03-22 15:19:07 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Use RTC_UNUSED instead of conditional compilation in BWE simulator tool.
2017-03-17 13:34:47 +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
…
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_send_stream.h
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
2017-03-30 17:44:38 +00:00
webrtc.gni
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +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.