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Henrik Kjellander a5073c0c97 Disable AudioDeviceTest.StartPlayoutOnTwoInstances on iOS 
This test fails when running on real iOS devices.

BUG=webrtc:6889
TBR=henrika@webrtc.org

Review-Url: https://codereview.webrtc.org/2576653002 .
Cr-Commit-Position: refs/heads/master@{#15586}
2016-12-14 06:25:24 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Fixing integer overflow when parsing bandwidth attribute.
2016-12-14 00:37:16 +00:00
audio
Support external audio mixer in webrtc 2.
2016-12-13 22:06:35 +00:00
base
Delete unused class AsyncFile.
2016-12-13 07:56:01 +00:00
build
Move /webrtc/api/android files to /webrtc/sdk/android
2016-12-06 12:29:45 +00:00
call
Disable failing perf test on Android.
2016-12-09 14:15:22 +00:00
common_audio
Update common_audio/smoothing_filter.
2016-12-13 14:53:07 +00:00
common_video
Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 14:08:39 +00:00
examples
Add a gtk3 port of peerconnection_client on Linux
2016-12-13 10:57:50 +00:00
libjingle
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
logging
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
media
Support external audio mixer in webrtc 2.
2016-12-13 22:06:35 +00:00
modules
Disable AudioDeviceTest.StartPlayoutOnTwoInstances on iOS
2016-12-14 06:25:24 +00:00
p2p
Fixing possible crash due to RefCountedChannel assignment operator.
2016-12-14 00:38:46 +00:00
pc
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
sdk
Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 14:08:39 +00:00
stats
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
system_wrappers
Add comment to metrics.h
2016-12-06 11:59:08 +00:00
test
Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 14:08:39 +00:00
tools
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
video
Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 14:08:39 +00:00
voice_engine
Delete voice_engine_configurations.h
2016-12-12 13:03:08 +00:00
.gitignore
…
BUILD.gn
Delete voice_engine_configurations.h
2016-12-12 13:03:08 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Revert of Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )
2016-12-09 10:35:30 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
typedefs.h
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 14:08:39 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +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.