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webrtc_m130/webrtc
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bjornv@webrtc.org f3e1341da7 VoEVolumeTest: Enabled Linux flaky tests 
Fixed error checks only on Linux to be able to turn on flaky tests. The cause of flaky failures is due to late values in pulse audio.

Related (deleted) CLs:
https://webrtc-codereview.appspot.com/19469007/
https://webrtc-codereview.appspot.com/19469004/

BUG=367
TESTED=trybots, voe_auto_test repeated
R=henrikg@webrtc.org, tina.legrand@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6195 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 10:43:42 +00:00
..
base
Remove IOKit linkage from iOS builds.
2014-05-19 21:05:10 +00:00
build
removed webrtc_base_tests_utils from merge libs as it was breaking some builds.
2014-05-15 21:45:09 +00:00
common_audio
common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16
2014-05-16 12:11:20 +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: couldn't run a second time. The reason is voe could register/unregister for each run, but vie would expect initialization only once per process.
2014-05-15 03:18:15 +00:00
modules
Add NACK and RPSI packet types to RTCP packet builder.
2014-05-20 09:53:51 +00:00
overrides/webrtc/base
Fix Windows debug compile of overrides/ logging.
2014-05-15 13:56:56 +00:00
system_wrappers
Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main}
2014-05-16 09:39:51 +00:00
test
Add NACK and RPSI packet types to RTCP packet builder.
2014-05-20 09:53:51 +00:00
tools
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
video
Fix Win VideoSendStream::...::ToString() compiles.
2014-05-15 10:03:24 +00:00
video_engine
Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main}
2014-05-16 09:39:51 +00:00
voice_engine
VoEVolumeTest: Enabled Linux flaky tests
2014-05-20 10:43:42 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Add DeliveryStatus enum to DeliverPacket().
2014-05-14 13:57:12 +00:00
common_types.h
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +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
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +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
Made the presubmit script accept license headers back to 2003
2014-05-15 18:21:17 +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
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
webrtc_examples.gyp
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
webrtc.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +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.