webrtc_m130

History

brettw@chromium.org 519c9e207d Convert GN visibility to be a list.

GN visibility currently allows either string or list types, but this is causing
some problems for some templates. I'm going to require it to be lists, so am
changing all callers before pushing the new binary.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7111 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-09-08 22:45:18 +00:00

base

GN: Prefix WebRTC specific variables with "rtc_"

2014-09-07 17:36:10 +00:00

build

GN: Prefix WebRTC specific variables with "rtc_"

2014-09-07 17:36:10 +00:00

common_audio

Add ctors to ChannelBuffer to enable copying on construction.

2014-09-08 20:27:04 +00:00

common_video

GN: Prefix WebRTC specific variables with "rtc_"

2014-09-07 17:36:10 +00:00

examples

Remove former team members from OWNERS and WATCHLISTS

2014-08-26 06:12:08 +00:00

libjingle/xmllite

Fix strange owners files with comments that crashs "git cl presubmit"

2014-09-04 14:25:07 +00:00

modules

Convert GN visibility to be a list.

2014-09-08 22:45:18 +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

sound

Add talk owners to migrated talk folders

2014-08-28 16:03:58 +00:00

system_wrappers

GN: Prefix WebRTC specific variables with "rtc_"

2014-09-07 17:36:10 +00:00

test

Drop buildbot_tests.py script

2014-09-05 08:25:38 +00:00

tools

Adding the ability to test on Chrome for Android.

2014-09-08 13:01:40 +00:00

video

Change return value for number of discarded packets to be int.

2014-09-04 07:07:44 +00:00

video_engine

Fix RTT calculations for send-only channels.

2014-09-08 08:45:25 +00:00

voice_engine

Partial revert of r7014 (Android APK refactor)

2014-09-03 14:34:46 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

BUILD.gn

GN: Prefix WebRTC specific variables with "rtc_"

2014-09-07 17:36:10 +00:00

call.h

Network up/down signaling in Call.

2014-09-03 16:17:12 +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

Roll chromium_revision 289723:291647

2014-08-25 14:16:32 +00:00

transport.h

Rename newapi::Transport::SendRTP()->SendRtp().

2013-11-20 12:17:04 +00:00

typedefs.h

Add CHECK and friends from Chromium.

2014-08-28 16:28:26 +00:00

video_engine_tests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

video_receive_stream.h

Change return value for number of discarded packets to be int.

2014-09-04 07:07:44 +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

Fix rm command for class cleanup in r7091

2014-09-06 22:11:28 +00:00

webrtc_perf_tests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

webrtc_tests.gypi

Partial revert of r7014 (Android APK refactor)

2014-09-03 14:34:46 +00:00

webrtc.gyp

Create a copy of talk/xmllite under webrtc/xmllite.

2014-09-02 15:41:12 +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.