webrtc

History

henrike@webrtc.org 88fbb2d86b Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. Same as https://webrtc-codereview.appspot.com/19519004. The issue in http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux... is solved by this change http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing... (tested locally). BUG=3380 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17619005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d		2014-05-21 21:18:46 +00:00
..
base	Rebase webrtc/base 6163:6216 (svn diff -r 6163:6216 http://webrtc.googlecode.com/svn/trunk/talk/base , apply diff manually)	2014-05-21 20:42:17 +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	Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.	2014-05-21 21:18:46 +00:00
common_video	Added include of assert.h for files calling assert but missing the include.	2014-04-29 17:54:17 +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	Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.	2014-05-21 21:18:46 +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	Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.	2014-05-21 21:18:46 +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	Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky.	2014-05-20 15:12:29 +00:00
video_engine	Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.	2014-05-21 21:18:46 +00:00
voice_engine	Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.	2014-05-21 21:18:46 +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	Made the presubmit script accept license headers back to 2003	2014-05-15 18:21:17 +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	Add ToString() to VideoSendStream::Config.	2014-05-15 09:35:06 +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 webrtc field trials API.	2014-05-14 12:24:04 +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.