webrtc

History

henrik.lundin@webrtc.org 36b6221cd4 Adding a link to issue BUG=3010 TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5668 4adac7df-926f-26a2-2b94-8c16560cd09d		2014-03-10 10:24:35 +00:00
..
build	Roll chromium_revision 249215:255773	2014-03-10 09:51:17 +00:00
common_audio	Add a deinterleaved float interface to AudioProcessing.	2014-03-04 20:58:13 +00:00
common_video	Add an AlignedFreeDeleter and remove scoped_ptr_malloc.	2014-02-20 21:08:36 +00:00
examples	Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera.	2014-01-24 18:42:12 +00:00
modules	Fix build breakage introduce with r5665.	2014-03-10 09:38:39 +00:00
system_wrappers	Remove std:: prefixes from C functions in webrtc/.	2014-03-07 15:23:34 +00:00
test	Re-enable libjingle_peerconnection_java_unittest since bug 2952 is fixed.	2014-03-01 00:07:08 +00:00
tools	Now printing less output from compare_videos.py.	2014-01-07 17:59:30 +00:00
video	Adding a link to issue	2014-03-10 10:24:35 +00:00
video_engine	Avoid crash in ViEEncoder::DeRegisterExternalEncoder().	2014-03-07 18:00:05 +00:00
voice_engine	Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.	2014-03-06 23:49:08 +00:00
.gitignore	.gitignore: Add *.mk, created as part of ChromiumOS build	2013-01-04 21:25:42 +00:00
call.h	Add configuration for cpu overuse detection to video send stream.	2014-01-31 10:05:07 +00:00
common_types.h	Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).	2014-02-19 11:59:02 +00:00
common.h	Add a Config class interface to AudioProcessing for passing options.	2013-07-25 18:28:29 +00:00
config.h	Wire up statistics in video receive stream of new API	2014-02-07 12:06:29 +00:00
engine_configurations.h	Remove external encryption API for VoE.	2014-02-18 11:27:22 +00:00
experiments.h	Missing include in experiments.h	2014-02-25 09:17:43 +00:00
frame_callback.h	Wire up statistics in video receive stream of new API	2014-02-07 12:06:29 +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
supplement.gypi	Removes script for generating supplement.gypi also adds git ignore for tools/gn.	2014-01-21 15:54:56 +00:00
transport.h	Rename newapi::Transport::SendRTP()->SendRtp().	2013-11-20 12:17:04 +00:00
typedefs.h	Moves WEBRTC_POSIX define from header file to gyp-settings.	2014-03-07 15:30:21 +00:00
video_engine_tests.isolate	Merge metrics_unittests into video_engine_tests.	2013-12-13 14:31:47 +00:00
video_receive_stream.h	Make VideoReceiveStream::GetStats() const.	2014-02-07 15:32:45 +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	Wire up statistics in video receive stream of new API	2014-02-07 12:06:29 +00:00
webrtc_examples.gyp	Android example apps: fixes issue where useful failure information was suppressed.	2014-01-21 19:03:51 +00:00
webrtc_perf_tests.isolate	Move realtime tests to webrtc_perf_tests.	2013-12-13 12:48:05 +00:00
webrtc_tests.gypi	Adding NetEq performance test to webrtc_perf_tests	2014-01-10 08:24:04 +00:00
webrtc.gyp	Integrate fake_network_pipe into direct_transport.	2013-12-18 20:28:25 +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.