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webrtc/webrtc
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solenberg@webrtc.org a28c697d93 - Get rid of 'using' from .h 
- Add parenthesis to make order of evaluation clearer.

BUG=
R=minyue@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6304 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-02 15:22:33 +00:00
..
base
Rebase webrtc/base with r6250:
2014-05-29 15:53:39 +00:00
build
removed webrtc_base_tests_utils from merge libs as it was breaking some builds.
2014-05-15 21:45:09 +00:00
common_audio
common_audio/signal_processing: Fixes arm compilation issues with gcc 4.8
2014-05-28 08:45:04 +00:00
common_video
Add support of texture frames for video capturer.
2014-05-28 07:00:51 +00:00
examples
WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker
2014-05-21 03:37:45 +00:00
modules
- Get rid of 'using' from .h
2014-06-02 15:22:33 +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
* Revert clock.cc changes made in 6178, but keep the changes to the test.
2014-05-29 19:40:28 +00:00
test
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
2014-05-21 21:18:46 +00:00
tools
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
2014-05-21 21:18:46 +00:00
video
Increase the threshold for CallPerfTest.CaptureNtpTimeWithNetworkDelay to avoid flaky.
2014-05-29 05:03:52 +00:00
video_engine
Add support of texture frames for video capturer.
2014-05-28 07:00:51 +00:00
voice_engine
Add a Reset() method to AudioFrame.
2014-05-30 17:28:50 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Add DeliveryStatus enum to DeliverPacket().
2014-05-14 13:57:12 +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
Merge metrics_unittests into video_engine_tests.
2013-12-13 14:31:47 +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
Revert "Revert "Remove VideoSendStreamInput::PutFrame.""
2014-05-23 13:03:45 +00:00
webrtc_examples.gyp
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Add native_test dependency to webrtc_perf_tests.
2014-05-30 12:42:10 +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.