platform-external-webrtc

Files

mflodman@webrtc.org f3973e81d5 Make sure channels in the same call are in the same channel group.

Tested manually. I'll make a follow CL with a proper test once review.webrtc.org/5619004 has been committed.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5280 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-12-13 09:40:45 +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

examples/android

Complete rewrite of demo application.

2013-12-12 16:55:37 +00:00

modules

Making RemoteRateControl::min_configured_bit_rate_ configurable

2013-12-13 08:42:42 +00:00

system_wrappers

Roll chromium_revision 232627:238260

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

test

Update talk to 58127566 together with

2013-12-13 00:21:03 +00:00

tools

Roll chromium_revision 232627:238260

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

video

Make sure channels in the same call are in the same channel group.

2013-12-13 09:40:45 +00:00

video_engine

Make sure channels in the same call are in the same channel group.

2013-12-13 09:40:45 +00:00

voice_engine

Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).

2013-12-11 21:42:44 +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

Update talk to 58127566 together with

2013-12-13 00:21:03 +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

Add SwapFrame() to VideoSendStreamInput.

2013-12-11 16:26:16 +00:00

webrtc_examples.gyp

Complete rewrite of demo application.

2013-12-12 16:55:37 +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.