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webrtc_m130/webrtc
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terelius f79dbadc09 Add has_value() and value() methods to rtc::Optional. 
These methods have the same behavior as their counterparts in std::optional, except that rtc::Optional::value() requires that the value exists whereas std::optional::value() throws an exception.

BUG=webrtc:7843

Review-Url: https://codereview.webrtc.org/2942203002
Cr-Commit-Position: refs/heads/master@{#18631}
2017-06-16 13:48:13 +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
Add has_value() and value() methods to rtc::Optional.
2017-06-16 13:48:13 +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
Remove unused #include "libyuv/compare.h"
2017-06-16 11:54:22 +00:00
modules
Reduces sensitivity in audio-glitch detector for iOS
2017-06-16 12:09:10 +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
Reduces sensitivity in audio-glitch detector for iOS
2017-06-16 12:09:10 +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
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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.