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henrik.lundin@webrtc.org 63e46077a3 Add thread annotations to parts of ACMGenericCodec 
This change adds annotations to all member variables that could be
annotated without acquiring any new locks, or changing the lock
structure in any other way.

BUG=3041
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6441 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 08:02:25 +00:00
..
base
Add missing sources to webrtc/base/base.gyp
2014-06-16 07:11:19 +00:00
build
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
common_audio
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
common_video
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
examples
AppRTCDemo(android): support app (UI) & capture rotation.
2014-06-06 18:40:44 +00:00
modules
Add thread annotations to parts of ACMGenericCodec
2014-06-16 08:02:25 +00:00
overrides/webrtc/base
Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.
2014-05-21 16:52:14 +00:00
system_wrappers
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
test
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
tools
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video
Enable pacing by default and remove the option to disable it from the new API.
2014-06-12 15:12:25 +00:00
video_engine
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
voice_engine
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
.gitignore
…
call.h
Make VideoSendStream/VideoReceiveStream configs const.
2014-06-06 10:49:19 +00:00
common_types.h
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
…
config.cc
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
config.h
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
engine_configurations.h
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
2014-05-12 12:19:19 +00:00
experiments.h
Adding API for setting bandwidth estimation configurations.
2014-03-25 10:37:31 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add kjellander@webrtc.org as OWNER for *.isolate
2014-06-10 05:42:53 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
supplement.gypi
Roll chromium_revision 260462:266514
2014-04-29 09:36:40 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Remove ALLOW_UNUSED.
2014-05-05 18:18:02 +00:00
video_engine_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video_receive_stream.h
Rename Start/Stop in Video{Send,Receive}Streams.
2014-04-24 11:13:21 +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
Enable pacing by default and remove the option to disable it from the new API.
2014-06-12 15:12:25 +00:00
webrtc_examples.gyp
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
Rename neteq4 folder to neteq
2014-06-09 08:10:28 +00:00
webrtc.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +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.