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kjellander@webrtc.org 55fcd716f3 Disable libjingle_peerconnection_java_unittest 
Broken by libjingle roll in r5590.

TBR=henrike@webrtc.org
BUG=2960
TEST=git try --bot=linux_baremetal --revision=5597

Review URL: https://webrtc-codereview.appspot.com/9029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5598 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-23 18:47:27 +00:00
..
build
Rename merged webrtc lib to libwebrtc_merged.a.
2014-02-11 18:22:29 +00:00
common_audio
Add an AlignedFreeDeleter and remove scoped_ptr_malloc.
2014-02-20 21:08:36 +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
Removed unused mock methods in audio_processing
2014-02-21 23:56:05 +00:00
system_wrappers
Add an AlignedFreeDeleter and remove scoped_ptr_malloc.
2014-02-20 21:08:36 +00:00
test
Disable libjingle_peerconnection_java_unittest
2014-02-23 18:47:27 +00:00
tools
Now printing less output from compare_videos.py.
2014-01-07 17:59:30 +00:00
video
Incorrect overhead calculation when using FEC + RTP extension headers.
2014-02-18 08:40:33 +00:00
video_engine
Updated WebRTC version to 3.50
2014-02-20 22:31:07 +00:00
voice_engine
Removes VoERTP_RTCP::InsertExtraRTPPacket.
2014-02-19 08:58:08 +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
Add experiment: SkipEncodingUnusedStreams
2014-02-10 09:20:51 +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
Added architecture for compiling under chrome NaCl.
2014-02-19 13:55:02 +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.