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mandermo a6069e8a01 Espresso test case to control loopback call 
The test case is put inside a new test target. That test target will be started from a test script to asses video quality.

BUG=webrtc:6545

Review-Url: https://codereview.webrtc.org/2585813002
Cr-Commit-Position: refs/heads/master@{#16088}
2017-01-16 10:23:09 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Added a new echo likelihood stat that reports the maximum value from a previous time period.
2017-01-15 16:29:46 +00:00
audio
Added a new echo likelihood stat that reports the maximum value from a previous time period.
2017-01-15 16:29:46 +00:00
base
Delete unused file faketaskrunner.h.
2017-01-16 08:30:33 +00:00
build
Remove lint ignores and add WrongCall ignore directly to the test.
2017-01-12 14:19:40 +00:00
call
Added a new echo likelihood stat that reports the maximum value from a previous time period.
2017-01-15 16:29:46 +00:00
common_audio
Replace RTC_DCHECK(false) with RTC_NOTREACHED().
2017-01-12 10:24:27 +00:00
common_video
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
examples
Espresso test case to control loopback call
2017-01-16 10:23:09 +00:00
logging
Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )
2017-01-13 14:52:12 +00:00
media
Added a new echo likelihood stat that reports the maximum value from a previous time period.
2017-01-15 16:29:46 +00:00
modules
Added a new echo likelihood stat that reports the maximum value from a previous time period.
2017-01-15 16:29:46 +00:00
p2p
Revert of Remove the IceTransportInternal2. (patchset #1 id:1 of https://codereview.webrtc.org/2632563002/ )
2017-01-16 08:12:52 +00:00
pc
Provide better message for when RTCP mux "require" policy is triggered.
2017-01-13 19:34:39 +00:00
sdk
Stop using deprecated constraints-based version of CreateAudioSource.
2017-01-13 19:47:56 +00:00
stats
Update smoothed bitrate.
2017-01-12 18:17:38 +00:00
system_wrappers
Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )
2017-01-09 09:22:16 +00:00
test
Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.
2017-01-13 15:41:19 +00:00
tools
Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )
2017-01-13 14:52:12 +00:00
video
Rename base/analytics/ to base/numerics/
2017-01-13 17:14:33 +00:00
voice_engine
Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )
2017-01-13 14:52:12 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Rename base/analytics/ to base/numerics/
2017-01-13 17:14:33 +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
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +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
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +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
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +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.