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henrik.lundin@webrtc.org 2fa17015d1 Re-enable NetEqExternalDecoderTest for Android 
The test runs without problems now.

BUG=3343
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16519005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6144 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 11:45:22 +00:00
..
base
Adds a modified copy of talk/base to webrtc/base. It is the first step in
2014-05-13 18:00:26 +00:00
build
Adds a modified copy of talk/base to webrtc/base. It is the first step in
2014-05-13 18:00:26 +00:00
common_audio
Fix constness of AudioBuffer accessors.
2014-04-30 16:44:13 +00:00
common_video
Added include of assert.h for files calling assert but missing the include.
2014-04-29 17:54:17 +00:00
examples
WebRTCDemo: correct set trace filter operation.
2014-05-05 04:24:47 +00:00
modules
Re-enable NetEqExternalDecoderTest for Android
2014-05-14 11:45:22 +00:00
overrides/webrtc/base
Adds a modified copy of talk/base to webrtc/base. It is the first step in
2014-05-13 18:00:26 +00:00
system_wrappers
Adds a modified copy of talk/base to webrtc/base. It is the first step in
2014-05-13 18:00:26 +00:00
test
Move gflags usage to video_loopback.
2014-05-13 11:26:40 +00:00
tools
Replace scoped_array<T> with scoped_ptr<T[]>.
2014-04-25 23:10:28 +00:00
video
Re-enable the BitrateEstimatorTest cases for the Call API.
2014-05-14 09:15:19 +00:00
video_engine
Remove WEBRTC_TRACE uses in video_engine/
2014-05-14 08:02:22 +00:00
voice_engine
VoEVolumeTest: Adds error return tests.
2014-05-14 09:03:33 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Remove TraceCallback use from Call.
2014-04-24 11:35:33 +00:00
common_types.h
Made common_types.h PacketTime declaration match https://code.google.com/p/webrtc/source/browse/trunk/talk/base/asyncpacketsocket.h#65
2014-04-29 17:50:47 +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
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
2014-05-12 12:19:19 +00:00
experiments.h
Adding API for setting bandwidth estimation configurations.
2014-03-25 10:37:31 +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
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
2014-04-14 20:08:03 +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
Roll chromium_revision 260462:266514
2014-04-29 09:36:40 +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
Merge metrics_unittests into video_engine_tests.
2013-12-13 14:31:47 +00:00
video_receive_stream.h
Rename Start/Stop in Video{Send,Receive}Streams.
2014-04-24 11:13:21 +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
Rename Start/Stop in Video{Send,Receive}Streams.
2014-04-24 11:13:21 +00:00
webrtc_examples.gyp
Make WebRTC Android examples build without sourcing envsetup.sh
2014-04-15 08:35:00 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Move gflags usage to video_loopback.
2014-05-13 11:26:40 +00:00
webrtc.gyp
Adds a modified copy of talk/base to webrtc/base. It is the first step in
2014-05-13 18:00:26 +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.