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stefan d7a418f93a Add an experiment for stricter pacing and ALR probing. 
BUG=webrtc:8072

Review-Url: https://codereview.webrtc.org/2994623002
Cr-Commit-Position: refs/heads/master@{#19270}
2017-08-08 13:51:05 +00:00
..
api
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
2017-08-01 06:22:01 +00:00
audio
Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
2017-07-26 14:28:51 +00:00
base
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
call
Reland of SSRC and RSID may only refer to one sink each in RtpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2993633002/ )
2017-08-04 13:34:54 +00:00
common_audio
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
common_video
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
examples
Update native plugin dll for turn servers and video.
2017-07-25 23:04:31 +00:00
logging
Use default header extension map in rtc_event_log2text
2017-08-08 12:40:26 +00:00
media
Renamed fields in common_types.h/RtcpStatistics.
2017-08-04 12:03:53 +00:00
modules
Add an experiment for stricter pacing and ALR probing.
2017-08-08 13:51:05 +00:00
ortc
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
2017-08-01 06:22:01 +00:00
p2p
Make Port (and subclasses) fully "Network"-based, instead of IP-based.
2017-08-04 22:01:57 +00:00
pc
Ignore "b=AS:-1" instead of treating as a hard error.
2017-08-04 00:49:30 +00:00
rtc_base
Make Port (and subclasses) fully "Network"-based, instead of IP-based.
2017-08-04 22:01:57 +00:00
rtc_tools
Use default header extension map in rtc_event_log2text
2017-08-08 12:40:26 +00:00
sdk
Adding deadbeef@ as owner of Objc-C PeerConnection-related headers.
2017-08-07 17:00:42 +00:00
stats
Make RTCStatsReport::ToString() return JSON-parseable string.
2017-07-28 14:29:12 +00:00
system_wrappers
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
test
Rename ViEEncoder to VideoStreamEncoder
2017-08-03 15:27:51 +00:00
video
Add an experiment for stricter pacing and ALR probing.
2017-08-08 13:51:05 +00:00
voice_engine
Renamed fields in common_types.h/RtcpStatistics.
2017-08-04 12:03:53 +00:00
.gitignore
…
BUILD.gn
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
codereview.settings
…
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Renamed fields in common_types.h/RtcpStatistics.
2017-08-04 12:03:53 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +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
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +00:00
webrtc.gni
Removing unused declared arg
2017-08-03 17:45:31 +00:00
whitespace.txt
…

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.