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ivoc fbb374d8ed Add a reliability term to the echo detector. 
This will ensure that the estimated likelihood starts at a low value and prevents initial spikes.

BUG=webrtc:6525

Review-Url: https://codereview.webrtc.org/2503843004
Cr-Commit-Position: refs/heads/master@{#15131}
2016-11-17 14:19:54 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
audio
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
base
Delete unused files httprequest.h and httprequest.cc.
2016-11-17 13:54:43 +00:00
build
MB: Remove configuration for unexisting bots.
2016-11-16 22:05:51 +00:00
call
Explicitly enable RED over RTX in rampup tests.
2016-11-17 12:18:42 +00:00
common_audio
Remove all references to GYP
2016-11-16 19:11:38 +00:00
common_video
Remove all references to GYP
2016-11-16 19:11:38 +00:00
examples
Propagate bitrate setting to RTCRtpSender.
2016-11-17 08:44:09 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
modules
Add a reliability term to the echo detector.
2016-11-17 14:19:54 +00:00
p2p
Remove all references to GYP
2016-11-16 19:11:38 +00:00
pc
Remove all references to GYP
2016-11-16 19:11:38 +00:00
sdk
Remove all references to GYP
2016-11-16 19:11:38 +00:00
stats
Remove all references to GYP
2016-11-16 19:11:38 +00:00
system_wrappers
Use NtpTime in RtcpMeasurement instead of uint sec/uint frac.
2016-11-17 10:27:20 +00:00
test
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
tools
Remove all references to GYP
2016-11-16 19:11:38 +00:00
video
Use NtpTime in RtcpMeasurement instead of uint sec/uint frac.
2016-11-17 10:27:20 +00:00
voice_engine
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Reduce full stack test time to 45 secs and add H264 and FlexFEC.
2016-11-17 07:57:00 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reland of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2491613005/ )
2016-11-17 07:23:15 +00:00
common_types.h
Add overhead per packet observer to the rtp_sender.
2016-11-17 09:38:48 +00:00
config.cc
Reland of Declare VideoCodec.codec_specific_info private (patchset #1 id:1 of https://codereview.webrtc.org/2491613005/ )
2016-11-17 07:23:15 +00:00
config.h
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Reland #2 of Issue 2434073003: Extract bitrate allocation ...
2016-11-16 15:41:45 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Remove RED/RTX workaround from sender/receiver and VideoEngine2.
2016-11-11 11:28:38 +00:00
video_send_stream.h
Wire up FlexfecSender in RTP module and VideoSendStream.
2016-11-15 13:25:44 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +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.