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webrtc_m130/webrtc
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bjornv@webrtc.org 721f970cba common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 
We should avoid macros in general (see style guide) and the shift ones are particular dangerous since they assume that the user apply a non-negative shift.

Related CL: https://webrtc-codereview.appspot.com/16669004

BUG=3348,3353
TESTED=trybots and manually on linux
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 10:30:14 +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
common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16
2014-06-16 10:30:14 +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
common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16
2014-06-16 10:30:14 +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
Implements start bitrate for new video API.
2014-06-16 08:57:39 +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
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Implements start bitrate for new video API.
2014-06-16 08:57:39 +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
…
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
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.