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henrik.lundin@webrtc.org a90abdef62 Add thread annotations to AcmReceiver 
This change adds thread annotations to AcmReceiver. These are the
annotations that could be added without changing acquiring the locks in
more locations, or changing the lock structure.

BUG=3401
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6376 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09 18:35:11 +00:00
..
base
Rebase webrtc/base with r6345 version of talk/base:
2014-06-05 20:40:11 +00:00
build
Make it possible to build webrtc for arm64.
2014-06-04 17:15:42 +00:00
common_audio
The correct fix of workaround in r6261.
2014-06-05 08:53:51 +00:00
common_video
Android: cleanup gtest_target_type conditions.
2014-06-04 20:46:50 +00:00
examples
AppRTCDemo(android): support app (UI) & capture rotation.
2014-06-06 18:40:44 +00:00
modules
Add thread annotations to AcmReceiver
2014-06-09 18:35:11 +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
Make some methods in Clock class const declared
2014-06-09 17:40:49 +00:00
test
Updated conformance tests and w3c-ified them.
2014-06-09 09:47:44 +00:00
tools
Android: cleanup gtest_target_type conditions.
2014-06-04 20:46:50 +00:00
video
Make VideoSendStream/VideoReceiveStream configs const.
2014-06-06 10:49:19 +00:00
video_engine
ViEAutoTestAndroid: Unbreak compile by casting void* to jobject.
2014-06-06 18:59:30 +00:00
voice_engine
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
2014-06-05 20:34:08 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
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
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
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
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
2014-04-14 20:08:03 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
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
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
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
Remove unused test_env.py from isolate files + fix nss path.
2014-06-09 14:35:09 +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
Make VideoSendStream/VideoReceiveStream configs const.
2014-06-06 10:49:19 +00:00
webrtc_examples.gyp
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
webrtc_perf_tests.isolate
Remove unused test_env.py from isolate files + fix nss path.
2014-06-09 14:35:09 +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.