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webrtc/webrtc
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henrike@webrtc.org 5654b305e5 Rebase webrtc/base with r6464 version of talk/base: 
cd webrtc/base
svn diff -r 6463:6464 http://webrtc.googlecode.com/svn/trunk/talk/base >
6464.diff
patch -p0 -i 6464.diff

BUG=3379
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12749005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6466 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 14:37:05 +00:00
..
base
Rebase webrtc/base with r6464 version of talk/base:
2014-06-17 14:37:05 +00:00
build
Initial GN work for WebRTC
2014-06-17 08:54:03 +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
Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."
2014-06-17 09:50:12 +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
Updated W3C getusermedia tests to the latest version of the spec.
2014-06-17 08:46:58 +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
Add max limit of number for overuses. When limit is reached always apply the rampup delay.
2014-06-16 14:27:19 +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
BUILD.gn
Initial GN work for WebRTC
2014-06-17 08:54:03 +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
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
Add kjellander@webrtc.org as OWNER for *.isolate
2014-06-10 05:42:53 +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
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.