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gnish 191113a46b Added implementation of four functions in the BBR congestion controller: 
1) Function responsible for receiving feedback, digesting data and deciding switch scenarios.
2) Function which enters Startup mode.
3) Function which exits Startup mode.
4) Function which calculates, whether or not full bandwidth is reached.

BUG=webrtc:7713

Review-Url: https://codereview.webrtc.org/2924603002
Cr-Commit-Position: refs/heads/master@{#18901}
2017-07-05 12:00:46 +00:00
..
api
Remove unused static VideoEncoder functions
2017-07-04 13:03:41 +00:00
audio
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
base
Delete SignalThread class.
2017-06-29 13:21:20 +00:00
call
Reland of Add received audio/video call duration metrics based on packets.
2017-07-04 11:05:06 +00:00
common_audio
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
common_video
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
examples
Fixes build issue based on usage of Android O specific API
2017-07-04 14:02:35 +00:00
logging
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
media
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
modules
Added implementation of four functions in the BBR congestion controller:
2017-07-05 12:00:46 +00:00
ortc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
p2p
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
pc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
rtc_base
Reland "Update includes for webrtc/{base => rtc_base} rename (3/3)"
2017-06-30 17:45:21 +00:00
rtc_tools
fix comment length
2017-07-03 07:47:44 +00:00
sdk
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
stats
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
system_wrappers
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
test
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
tools
Revert of Remove webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2970743003/ )
2017-07-04 09:02:49 +00:00
video
Fix issue with zero rtt reports when using FlexFEC and add perf test.
2017-07-05 10:03:02 +00:00
voice_engine
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
.gitignore
…
BUILD.gn
Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ )
2017-06-30 15:11:44 +00:00
codereview.settings
…
common_types.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
common_types.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
config_unittest.cc
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
config.cc
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
config.h
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +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
…
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
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
video_send_stream.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +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.