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kthelgason b13237b9da Fix deprecated methods in AppRTCMobile. 
Remove warning suppressions and update code that uses deprecated methods
to the currently supported APIs.

BUG=webrtc:5549

Review-Url: https://codereview.webrtc.org/2780433006
Cr-Commit-Position: refs/heads/master@{#17472}
2017-03-30 11:56:05 +00:00
..
api
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
audio
Add Windows, Mac, Android support to low bandwidth audio test
2017-03-30 11:01:30 +00:00
base
To accommodate some downstream WebRTC users we need to loosen
2017-03-30 08:24:20 +00:00
call
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
common_audio
iSAC fix entropy coder: Recently added DCHECK could in fact trigger
2017-03-13 12:28:47 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Fix deprecated methods in AppRTCMobile.
2017-03-30 11:56:05 +00:00
logging
To accommodate some downstream WebRTC users we need to loosen
2017-03-30 08:24:20 +00:00
media
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
2017-03-30 06:57:43 +00:00
modules
To accommodate some downstream WebRTC users we need to loosen
2017-03-30 08:24:20 +00:00
ortc
Disable the ORTC integration tests on TSan.
2017-03-28 20:45:18 +00:00
p2p
Disable possibly flaky AudioCodingModuleTest.TestPacketLossStereo for Linux.
2017-03-28 10:00:07 +00:00
pc
Rewrite PeerConnection integration tests using better testing practices.
2017-03-30 04:08:16 +00:00
sdk
Fix JNI reference leak in MediaCodecVideoDecoder
2017-03-30 08:08:36 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
test
Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
2017-03-30 06:57:43 +00:00
tools
To accommodate some downstream WebRTC users we need to loosen
2017-03-30 08:24:20 +00:00
video
Disable flaky test EndToEndTest.TriggerMidCallProbing
2017-03-30 11:24:08 +00:00
voice_engine
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
.gitignore
…
BUILD.gn
Add Windows, Mac, Android support to low bandwidth audio test
2017-03-30 11:01:30 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Delete unused constant kConfigParameterSize.
2017-03-22 15:19:07 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +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
…
typedefs.h
Use RTC_UNUSED instead of conditional compilation in BWE simulator tool.
2017-03-17 13:34:47 +00:00
video_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_send_stream.h
Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
2017-03-21 20:26:06 +00:00
webrtc.gni
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.