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pbos@webrtc.org 391b4db7de Fix common_video_unittests in apk_tests.gyp. 
r5265 moved common_video_unittests to its own gyp, this required an
update of apk_tests.gyp that wasn't caught by our trybots.

TBR=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/5629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5266 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 16:48:53 +00:00
..
build
Fix common_video_unittests in apk_tests.gyp.
2013-12-11 16:48:53 +00:00
common_audio
Roll chromium_revision 232627:238260
2013-12-11 13:37:12 +00:00
common_video
Add SwapFrame() to VideoSendStreamInput.
2013-12-11 16:26:16 +00:00
modules
Add SwapFrame() to VideoSendStreamInput.
2013-12-11 16:26:16 +00:00
system_wrappers
Roll chromium_revision 232627:238260
2013-12-11 13:37:12 +00:00
test
Add SwapFrame() to VideoSendStreamInput.
2013-12-11 16:26:16 +00:00
tools
Roll chromium_revision 232627:238260
2013-12-11 13:37:12 +00:00
video
Add SwapFrame() to VideoSendStreamInput.
2013-12-11 16:26:16 +00:00
video_engine
Add SwapFrame() to VideoSendStreamInput.
2013-12-11 16:26:16 +00:00
voice_engine
Roll chromium_revision 232627:238260
2013-12-11 13:37:12 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Adds support for sending redundant payloads over RTX.
2013-12-04 10:24:26 +00:00
common_types.h
Revert 5259 "Callback for send bitrate estimates"
2013-12-11 14:07:33 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.h
Rename RTP-extension constants.
2013-11-20 11:48:56 +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
experiments.h
Adds support for sending redundant payloads over RTX.
2013-12-04 10:24:26 +00:00
frame_callback.h
Implement and test EncodedImageCallback in new ViE API.
2013-11-26 11:41:59 +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
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
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Correctly define OVERRIDE when building with g++ 4.7 and C++11 support
2013-12-10 19:20:46 +00:00
video_engine_tests.isolate
Roll chromium_revision 232627:238260
2013-12-11 13:37:12 +00:00
video_receive_stream.h
Set local SSRC for VideoReceiveStream.
2013-12-03 10:13:04 +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 SwapFrame() to VideoSendStreamInput.
2013-12-11 16:26:16 +00:00
webrtc_tests.gypi
Remove CallTest dependency on voice_engine/test/.
2013-12-05 14:24:17 +00:00
webrtc.gyp
Add SwapFrame() to VideoSendStreamInput.
2013-12-11 16:26:16 +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.