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oprypin 054be4c9d9 Support running AppRTC without a TURN ICE server 
The "ice_server_url" key is always present in AppRTC server's JSON response but may be empty if no ICE server is wanted.

BUG=webrtc:7636

Review-Url: https://codereview.webrtc.org/2878673002
Cr-Commit-Position: refs/heads/master@{#18118}
2017-05-12 07:40:58 +00:00
..
api
Reland of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2877013002/ )
2017-05-12 05:07:37 +00:00
audio
Fixing video loopback test with encoder factory.
2017-05-10 11:06:11 +00:00
base
Delete left-over declaration of AdjustCurrentProcessPrivilege.
2017-05-10 13:04:45 +00:00
call
Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.
2017-05-11 15:00:58 +00:00
common_audio
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
2017-04-28 12:24:50 +00:00
common_video
Add support for I444 in VideoFrameBuffer
2017-05-11 12:11:57 +00:00
examples
Support running AppRTC without a TURN ICE server
2017-05-12 07:40:58 +00:00
logging
DCHECK that we don't insert nullptr into event log.
2017-05-11 13:05:27 +00:00
media
WebRtcVideoEncoderFactory cleanup
2017-05-12 07:24:40 +00:00
modules
Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.
2017-05-11 15:00:58 +00:00
ortc
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
p2p
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
pc
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
sdk
WebRtcVideoEncoderFactory cleanup
2017-05-12 07:24:40 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
2017-04-28 12:24:50 +00:00
test
Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.
2017-05-11 15:00:58 +00:00
tools
Moving scripts to download and build apprtc/collider.
2017-05-09 07:13:47 +00:00
video
Make Call::OnRecoveredPacket parse RTP header and call OnRtpPacket.
2017-05-11 15:00:58 +00:00
voice_engine
Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend()
2017-05-03 15:30:15 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Remove config.h from //webrtc:webrtc target.
2017-05-03 12:28:46 +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
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Delete RawVideoType enum, use the VideoType enum instead.
2017-04-28 14:18:05 +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
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
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
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
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 Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )
2017-05-10 12:25:59 +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.