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Anders Carlsson e150058236 Move setting switches in AppRTCMobile to Settings screen 
All setting switches except "Loopback mode" is now in the Settings
screen instead of the main screen. They are also persisted across app
launches.

Bug: webrtc:7748
Change-Id: Iafd84e5e39639770118e2503148d1bf7fb9c3d8d
Reviewed-on: https://chromium-review.googlesource.com/527034
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18626}
2017-06-16 11:16:19 +00:00
..
api
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
audio
Also scan stderr for audio files to test, due to change in Android test_runner
2017-06-14 16:35:11 +00:00
base
Delete SignalSrtpError.
2017-06-16 07:57:21 +00:00
call
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
common_audio
Moving compile_assert_c.h to webrtc/base
2017-05-15 15:06:47 +00:00
common_video
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
examples
Move setting switches in AppRTCMobile to Settings screen
2017-06-16 11:16:19 +00:00
logging
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
media
Delete SignalSrtpError.
2017-06-16 07:57:21 +00:00
modules
Validate references of frames inserted into FrameBuffer2.
2017-06-15 16:06:21 +00:00
ortc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
p2p
Adding PortAllocator option to support cases where sockets can't be bound.
2017-06-13 22:49:45 +00:00
pc
Delete SignalSrtpError.
2017-06-16 07:57:21 +00:00
sdk
Create AndroidVideoBuffer and allow renderers to consume it.
2017-06-16 08:31:37 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Prevent data race in GetStaticInstance
2017-05-29 07:01:14 +00:00
test
Fix Chromium style checker warnings for MockAudioDecoder
2017-06-14 10:31:17 +00:00
tools
Add NetEq delay plotting to event_log_visualizer
2017-06-14 13:09:58 +00:00
video
Use the same QP max for tests as in production
2017-06-15 13:34:42 +00:00
voice_engine
Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:
2017-06-12 19:45:32 +00:00
.gitignore
…
BUILD.gn
Delete old include file webrtc/video_frame.h.
2017-06-13 13:05:05 +00:00
codereview.settings
…
common_types.cc
Add RSID-based demuxing to RtpDemuxer
2017-06-08 11:19:13 +00:00
common_types.h
Add RSID-based demuxing to RtpDemuxer
2017-06-08 11:19:13 +00:00
config.cc
Fix default extension id for video content type extension
2017-04-26 09:35:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Delete old include file webrtc/video_frame.h.
2017-06-13 13:05:05 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Add field trial for balanced degradation preference.
2017-06-14 12:52:21 +00:00
webrtc.gni
Adds detection of audio glitches for playout on iOS (reland)
2017-06-13 16:00:18 +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.