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gnish a36165c77b Final version of BBR, with tweaks made for WebRTC, major changes: 
1) Entering PROBE_RTT when necessary.
2) Congestion window gain of 0.65 instead of constant 4 packets.
3) {1.1, 0.9} pair instead of {1.25, 0.75}
4) Recovery mode.
5) No reaction to losses due to Recovery mode's implementation.
6) Supports encoder.
7) A new test compiling most of the simulation tests.
8) Bucket for high gain phase, disabled by default.
9) Pacer specific to BBR.

BUG=webrtc:7713

Review-Url: https://codereview.webrtc.org/2999073002
Cr-Commit-Position: refs/heads/master@{#19418}
2017-08-20 16:19:58 +00:00
..
api
iSAC fixed-point implementation of the Audio{En,De}coderFactoryTemplate APIs
2017-08-18 11:09:40 +00:00
audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
call
Reduce code repetition in RtcpDemuxerTest.
2017-08-18 18:41:39 +00:00
common_audio
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
common_video
Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )
2017-08-18 09:51:12 +00:00
examples
Remove dep on system_wrappers for apprtc and use only framework.
2017-08-16 14:28:03 +00:00
logging
Add rtc_event_log_unittest_helper.h to relevant BUILD.gn
2017-08-17 15:32:39 +00:00
media
Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings.
2017-08-17 15:58:54 +00:00
modules
Final version of BBR, with tweaks made for WebRTC, major changes:
2017-08-20 16:19:58 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Add TLS TURN tests.
2017-08-17 23:03:04 +00:00
pc
Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer.
2017-08-17 21:10:50 +00:00
rtc_base
Fix a delete type mismatch (deleting a sockaddr_in6* as a sockaddr*)
2017-08-18 07:15:19 +00:00
rtc_tools
Make path to wav file for jitter buffer simulation in event_log_visualizer configurable.
2017-08-16 09:48:33 +00:00
sdk
Fix a bug where listener custom drawer was ignored in EglRenderer.
2017-08-18 18:38:32 +00:00
stats
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
system_wrappers
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +00:00
test
Add Jpeg frame writer for test support.
2017-08-18 16:00:04 +00:00
video
Add Jpeg frame writer for test support.
2017-08-18 16:00:04 +00:00
voice_engine
Revert of Stop silently accepting unsupported flags in test binaries (patchset #5 id:150001 of https://codereview.webrtc.org/2968003003/ )
2017-08-18 14:22:36 +00:00
.gitignore
…
BUILD.gn
Reland of Make it possible for tests to set up trace event handlers. (patchset #1 id:1 of https://codereview.webrtc.org/2997963002/ )
2017-08-17 14:59:54 +00:00
codereview.settings
…
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
config.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +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
Replace absolute path with relative path for GN files.
2017-08-15 15:57:36 +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.