platform-external-webrtc

Files

kjellander@webrtc.org f9bdbe3619 Roll chromium_revision 232627:238260

This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003

TEST=trybots passing
BUG=none
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-12-11 13:37:12 +00:00

build

Roll chromium_revision 232627:238260

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common_audio

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

common_video

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

modules

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

system_wrappers

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

test

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

tools

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

video

Improve VideoSendStreamTest::MaxPacketSize

2013-12-10 10:05:17 +00:00

video_engine

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +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

Callback for send bitrate estimates

2013-12-11 12:53:03 +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

Implement and test EncodedImageCallback in new ViE API.

2013-11-26 11:41:59 +00:00

webrtc_tests.gypi

Remove CallTest dependency on voice_engine/test/.

2013-12-05 14:24:17 +00:00

webrtc.gyp

Move implementation files out of the webrtc/ root.

2013-12-05 12:11:47 +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.