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pbos@webrtc.org 53c8573525 Rename video streams' start/stop methods. 
{Start,Stop}{Send,Receive}() -> {Start,Stop}{Sending,Receiving}().

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5136 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 11:36:47 +00:00
..
build
Remove ".." from include_dirs in build/common.
2013-11-11 10:20:27 +00:00
common_audio
Upgrade scoped_ptr to Chromium's latest version.
2013-10-22 12:50:00 +00:00
common_video
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
2013-10-31 23:41:04 +00:00
modules
Add experimental noise suppression dummy API.
2013-11-19 15:17:51 +00:00
system_wrappers
Remove ".." from include_dirs in build/common.
2013-11-11 10:20:27 +00:00
test
Fix test broken with r5128.
2013-11-18 11:58:24 +00:00
tools
Remove update_resources.py as it's no longer used.
2013-11-08 09:08:36 +00:00
video
Rename video streams' start/stop methods.
2013-11-20 11:36:47 +00:00
video_engine
Rename AutoMute to SuspendBelowMinBitrate
2013-11-18 12:18:43 +00:00
voice_engine
Add experimental noise suppression dummy API.
2013-11-19 15:17:51 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call_tests.cc
Rename video streams' start/stop methods.
2013-11-20 11:36:47 +00:00
call.cc
Rename Call::Create{Receive,Send}Stream().
2013-11-20 10:40:25 +00:00
call.h
Rename Call::Create{Receive,Send}Stream().
2013-11-20 10:40:25 +00:00
common_types.h
Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.
2013-10-28 09:21:07 +00:00
common_unittest.cc
Fix unnamed-type-template-args warnings on clang.
2013-06-27 10:18:09 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
engine_configurations.h
Adds a new voice engine warning for the typing noise off state.
2013-09-17 18:09:20 +00:00
frame_callback.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +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
loopback.cc
Rename video streams' start/stop methods.
2013-11-20 11:36:47 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.
2013-07-22 22:32:50 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
transport.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
typedefs.h
Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined.
2013-10-23 19:11:32 +00:00
video_engine_tests.isolate
Hook up audio/video sync to Call.
2013-11-18 11:45:11 +00:00
video_receive_stream.h
Rename video streams' start/stop methods.
2013-11-20 11:36:47 +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
Rename video streams' start/stop methods.
2013-11-20 11:36:47 +00:00
webrtc_tests.gypi
Hook up audio/video sync to Call.
2013-11-18 11:45:11 +00:00
webrtc.gyp
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +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.