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philipp.hancke 6275dfdd06 Fix comment about remote restart being requested in createOffer 
BUG=None

Review-Url: https://codereview.webrtc.org/2837153002
Cr-Commit-Position: refs/heads/master@{#17870}
2017-04-25 19:57:14 +00:00
..
api
GN: Tighten up test target visibility + refactorings
2017-04-25 11:04:50 +00:00
audio
GN: Tighten up test target visibility + refactorings
2017-04-25 11:04:50 +00:00
base
Add SafeMin() and SafeMax(), which accept args of different types
2017-04-25 19:35:07 +00:00
call
GN: Tighten up test target visibility + refactorings
2017-04-25 11:04:50 +00:00
common_audio
Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ )
2017-04-24 19:26:27 +00:00
common_video
Add SafeMin() and SafeMax(), which accept args of different types
2017-04-25 19:35:07 +00:00
examples
Remove the expat and jsoncpp dependencies to reduce binary size.
2017-04-24 19:05:06 +00:00
logging
Delay based logging.
2017-04-11 08:50:23 +00:00
media
Enabling 'gn check' on webrtc/video.
2017-04-25 07:35:35 +00:00
modules
Add SafeMin() and SafeMax(), which accept args of different types
2017-04-25 19:35:07 +00:00
ortc
Enabling 'gn check' on webrtc/video.
2017-04-25 07:35:35 +00:00
p2p
GN: Tighten up test target visibility + refactorings
2017-04-25 11:04:50 +00:00
pc
Fix comment about remote restart being requested in createOffer
2017-04-25 19:57:14 +00:00
sdk
Android: Move EglBase10/EglBase14 from to API to src
2017-04-25 12:32:07 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ )
2017-04-24 19:26:27 +00:00
test
GN: Tighten up test target visibility + refactorings
2017-04-25 11:04:50 +00:00
tools
GN: Tighten up test target visibility + refactorings
2017-04-25 11:04:50 +00:00
video
GN: Tighten up test target visibility + refactorings
2017-04-25 11:04:50 +00:00
voice_engine
Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ )
2017-04-24 19:26:27 +00:00
.gitignore
…
BUILD.gn
GN: Tighten up test target visibility + refactorings
2017-04-25 11:04:50 +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
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
config.cc
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
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
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +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
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_frame.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
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Add number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +00:00
webrtc.gni
Remove the expat and jsoncpp dependencies to reduce binary size.
2017-04-24 19:05:06 +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.