webrtc

History

pbos@webrtc.org 5a63655ab0 Rename Call::Create{Receive,Send}Stream().

Renaming the methods to include Video. Long-term there will hopefully be
AudioSendStream/AudioReceiveStreams as well.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5135 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-11-20 10:40:25 +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 Call::Create{Receive,Send}Stream().

2013-11-20 10:40:25 +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 Call::Create{Receive,Send}Stream().

2013-11-20 10:40:25 +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 Call::Create{Receive,Send}Stream().

2013-11-20 10:40:25 +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

Separate Call API/build files from video_engine/.

2013-10-28 16:32:01 +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 AutoMute to SuspendBelowMinBitrate

2013-11-18 12:18:43 +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.