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kjellander@webrtc.org 49fa212bcd Fix compile error on JDK 1.7. 
JDK 1.7 gives an error like this:
warning: [static] static method should be qualified by type name

R=pbos@webrtc.org
TBR=henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/29399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7133 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 12:35:59 +00:00
..
base
include cstdlib for free() and abort()
2014-09-10 03:24:36 +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
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +00:00
examples
Fix compile error on JDK 1.7.
2014-09-10 12:35:59 +00:00
libjingle/xmllite
Fix strange owners files with comments that crashs "git cl presubmit"
2014-09-04 14:25:07 +00:00
modules
Remove DestructEncoderInst and its codec-specific implementations.
2014-09-10 08:52:26 +00:00
overrides/webrtc/base
…
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
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +00:00
tools
Bot Browser files moved to /bot/browser/
2014-09-09 14:50:09 +00:00
video
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +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
…
BUILD.gn
Fix the RTC+Chromium GN build.
2014-09-09 19:15:33 +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
…
common.h
…
config.cc
…
config.h
Check before send/receive rtp header extensions.
2014-07-20 15:27:35 +00:00
engine_configurations.h
…
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
…
typedefs.h
Add CHECK and friends from Chromium.
2014-08-28 16:28:26 +00:00
video_encoder.h
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +00:00
video_engine_tests.isolate
…
video_frame.h
Expose VideoEncoders with webrtc/video_encoder.h.
2014-09-09 10:40:56 +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
…
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
…
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.