webrtc

History

minyue@webrtc.org 194fea7640 The lastest commit on this file was in https://webrtc-codereview.appspot.com/15529004/ The final patch set should have included this, but was missed. R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6755 4adac7df-926f-26a2-2b94-8c16560cd09d		2014-07-22 09:55:51 +00:00
..
base	Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.	2014-07-16 21:28:26 +00:00
build	Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.	2014-07-11 13:44:02 +00:00
common_audio	int16<->float conversions: Use size_t for array length argument, not int	2014-07-16 08:36:52 +00:00
common_video	GN: Implement BUILD.gn for common_video.	2014-07-03 17:04:12 +00:00
examples	WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering.	2014-07-03 05:59:22 +00:00
modules	The lastest commit on this file was in	2014-07-22 09:55:51 +00:00
overrides/webrtc/base	Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.	2014-05-21 16:52:14 +00:00
system_wrappers	Sleep in ThreadTest thread functions.	2014-07-18 10:12:50 +00:00
test	Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams.	2014-07-10 10:35:12 +00:00
tools	Pass GYP DEPTH variable to isolate.	2014-06-13 09:02:15 +00:00
video	Check before send/receive rtp header extensions.	2014-07-20 15:27:35 +00:00
video_engine	Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.	2014-07-16 21:28:26 +00:00
voice_engine	This is related to an earlier CL of enabling Opus 48 kHz.	2014-07-18 12:28:28 +00:00
.gitignore	.gitignore: Add *.mk, created as part of ChromiumOS build	2013-01-04 21:25:42 +00:00
BUILD.gn	GN: Fix include paths for WebRTC in Chromium build.	2014-07-13 09:02:54 +00:00
call.h	Remove GetDefaultConfigs() from Call.	2014-07-07 04:45:15 +00:00
common_types.h	Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.	2014-07-11 13:44:02 +00:00
common.gyp	Add ToString() to VideoSendStream::Config.	2014-05-15 09:35:06 +00:00
common.h	Add a Config class interface to AudioProcessing for passing options.	2013-07-25 18:28:29 +00:00
config.cc	Add ToString() to VideoSendStream::Config.	2014-05-15 09:35:06 +00:00
config.h	Check before send/receive rtp header extensions.	2014-07-20 15:27:35 +00:00
engine_configurations.h	Add boilerplate code for H.264.	2014-07-04 12:42:07 +00:00
experiments.h	Remove no longer used SkipEncodingUnusedStreams.	2014-07-22 07:17:17 +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
OWNERS	GN: Add BUILD.gn files + kjellander to OWNERS	2014-06-23 19:21:07 +00:00
PATENTS	Move src/ -> webrtc/	2012-10-22 18:19:23 +00:00
PRESUBMIT.py	PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.	2014-05-23 17:27:18 +00:00
README.chromium	Move src/ -> webrtc/	2012-10-22 18:19:23 +00:00
supplement.gypi	TSan: Move suppressions to source file.	2014-06-27 09:18:51 +00:00
transport.h	Rename newapi::Transport::SendRTP()->SendRtp().	2013-11-20 12:17:04 +00:00
typedefs.h	Remove ALLOW_UNUSED.	2014-05-05 18:18:02 +00:00
video_engine_tests.isolate	Pass GYP DEPTH variable to isolate.	2014-06-13 09:02:15 +00:00
video_receive_stream.h	Remove GetDefaultConfigs() from Call.	2014-07-07 04:45:15 +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	Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.	2014-07-11 13:44:02 +00:00
webrtc_examples.gyp	WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering.	2014-07-03 05:59:22 +00:00
webrtc_perf_tests.isolate	Pass GYP DEPTH variable to isolate.	2014-06-13 09:02:15 +00:00
webrtc_tests.gypi	Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable.	2014-07-08 13:59:46 +00:00
webrtc.gyp	Add ToString() to VideoSendStream::Config.	2014-05-15 09:35:06 +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.