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andrew@webrtc.org cc476aa038 Fix a name collision with Android libc++ 
The Android libc++ has a symbol called '_P'
This CL renames a property called _P in webrtc.

BUG=chromium:427718
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30009004

Patch from Fabrice de Gans-Riberi <fdegans@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7579 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 16:01:25 +00:00
..
base
Only configure the SSL library in one place.
2014-10-27 18:13:40 +00:00
build
arm64 iOS build.
2014-10-31 00:14:39 +00:00
common_audio
Revert "Revert part of r7561, "Refactor audio conversion functions.""
2014-10-31 04:58:14 +00:00
common_video
GN: Add common configs to all targets.
2014-09-28 17:37:22 +00:00
examples
Split video engine android initialization into each internal module initialization.
2014-09-17 11:44:51 +00:00
libjingle
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
modules
Simplify bwe tests.
2014-10-31 10:47:12 +00:00
overrides
webrtc/overrides: add OWNERS-file.
2014-09-17 08:04:28 +00:00
p2p
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Fix a name collision with Android libc++
2014-10-31 16:01:25 +00:00
test
Use external VideoDecoders in VideoReceiveStream.
2014-10-29 15:28:39 +00:00
tools
Adding the subtool rtcBot report visualizer
2014-10-24 09:26:16 +00:00
video
Implement conference-mode temporal-layer screencast.
2014-10-31 13:08:10 +00:00
video_engine
Remove unused code in overuse detector.
2014-10-29 10:05:21 +00:00
voice_engine
Add a simple AudioConverter class.
2014-10-27 18:18:17 +00:00
.gitignore
…
BUILD.gn
arm64 iOS build.
2014-10-31 00:14:39 +00:00
call.h
Remove -1 from Call::Config::start_bitrate_bps.
2014-10-14 11:52:10 +00:00
common_types.h
Add stats for duplicate sent and received NACK requests.
2014-10-29 12:42:30 +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
Implement conference-mode temporal-layer screencast.
2014-10-31 13:08:10 +00:00
config.h
Implement conference-mode temporal-layer screencast.
2014-10-31 13:08:10 +00:00
engine_configurations.h
Enable G.722 for Chromium builds
2014-10-29 08:32:44 +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
…
LICENSE_THIRD_PARTY
…
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
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
…
rtc_unittests.isolate
Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest.
2014-09-30 14:21:10 +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_decoder.h
Use external VideoDecoders in VideoReceiveStream.
2014-10-29 15:28:39 +00:00
video_encoder.h
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
2014-10-17 18:54:46 +00:00
video_engine_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video_frame.h
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
2014-09-17 09:02:25 +00:00
video_receive_stream.h
Configure A/V sync in WebRtcVideoEngine2.
2014-10-31 12:59:34 +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 min transmit bitrate to VideoEncoderConfig.
2014-10-24 09:23:21 +00:00
webrtc_examples.gyp
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
webrtc.gyp
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +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.