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alessiob aa7d935cd5 Evaluation scores export library and CSS file for the exported HTML file 
BUG=webrtc:7218
NOTRY=True

Review-Url: https://codereview.webrtc.org/2723703002
Cr-Commit-Position: refs/heads/master@{#17526}
2017-04-04 09:58:09 +00:00
..
api
Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ )
2017-04-03 16:15:52 +00:00
audio
Let PacketRouter separate send and receive modules.
2017-03-31 12:44:52 +00:00
base
Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
2017-03-31 10:08:07 +00:00
call
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
2017-04-03 06:53:04 +00:00
common_audio
iSAC fix entropy coder: Recently added DCHECK could in fact trigger
2017-03-13 12:28:47 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Reland of Enable trendline experiment and bayesian bitrate estimator experiment by default.
2017-04-01 13:51:09 +00:00
logging
Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
2017-03-31 10:08:07 +00:00
media
Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ )
2017-04-03 16:15:52 +00:00
modules
Evaluation scores export library and CSS file for the exported HTML file
2017-04-04 09:58:09 +00:00
ortc
Disable the ORTC integration tests on TSan.
2017-03-28 20:45:18 +00:00
p2p
Disable possibly flaky AudioCodingModuleTest.TestPacketLossStereo for Linux.
2017-03-28 10:00:07 +00:00
pc
Fixing some case-sensitive codec name comparisons.
2017-04-03 17:03:35 +00:00
sdk
[iOS] Added an initialization method to RTCConfiguration that takes a
2017-04-03 22:06:37 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Revert of Add empty header to fix internal project. (patchset #1 id:1 of https://codereview.webrtc.org/2790493006/ )
2017-04-03 09:33:33 +00:00
test
Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ )
2017-04-03 16:15:52 +00:00
tools
Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
2017-03-31 10:08:07 +00:00
video
Fixing some case-sensitive codec name comparisons.
2017-04-03 17:03:35 +00:00
voice_engine
Revert of Supporting 48kHz PCM file. (patchset #1 id:1 of https://codereview.webrtc.org/2790493004/ )
2017-03-31 23:32:28 +00:00
.gitignore
…
BUILD.gn
Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
2017-03-31 10:08:07 +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
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Revert of Supporting 48kHz PCM file. (patchset #1 id:1 of https://codereview.webrtc.org/2790493004/ )
2017-03-31 23:32:28 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
DEPS
Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ )
2017-04-03 16:15:52 +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
Use RTC_UNUSED instead of conditional compilation in BWE simulator tool.
2017-03-17 13:34:47 +00:00
video_decoder.h
Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ )
2017-04-03 16:15:52 +00:00
video_encoder.h
Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ )
2017-04-03 16:15:52 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +00:00
video_send_stream.h
Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
2017-04-03 06:53:04 +00:00
webrtc.gni
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +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.