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ivoc 9f4a4a096b Add empty residual echo detector. 
This CL does not contain the actual algorithm, but only creates an empty processing component and connects the right signals to it. The algorithm will be added in a follow-up CL.

BUG=webrtc:6525

Review-Url: https://codereview.webrtc.org/2405403003
Cr-Commit-Position: refs/heads/master@{#14820}
2016-10-28 12:39:23 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Move RTCStatsCollector helper functions to anonymous namespace.
2016-10-28 12:14:58 +00:00
audio
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
base
Moved rtc::Base64 to base approved.
2016-10-28 12:23:25 +00:00
build
Restore symbol level for Android builds.
2016-10-27 07:00:20 +00:00
call
Remove use of VoECodec in video/call tests.
2016-10-27 07:23:14 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Reland of Move bitstream parser to more appropriate directory. (patchset #1 id:1 of https://codereview.webrtc.org/2430353004/ )
2016-10-26 09:48:24 +00:00
examples
Add UINavigationController and settings bar button to AppRTCMobile.
2016-10-28 11:53:20 +00:00
libjingle
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
logging
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
media
Revert of Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #4 id:60001 of https://codereview.webrtc.org/2455063002/ )
2016-10-28 11:48:53 +00:00
modules
Add empty residual echo detector.
2016-10-28 12:39:23 +00:00
p2p
Fix "IsLoopbackIp" to cover all loopback addresses; not just 127.0.0.1.
2016-10-28 01:30:28 +00:00
pc
Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit.
2016-10-25 17:50:41 +00:00
sdk
iOS: Optimize video scaling and cropping
2016-10-20 10:34:32 +00:00
stats
RTCStats equality operator added.
2016-10-25 11:31:27 +00:00
system_wrappers
Delete DataLog abstraction, which was almost unused.
2016-10-21 08:53:01 +00:00
test
Remove use of VoECodec in video/call tests.
2016-10-27 07:23:14 +00:00
tools
Use bayesian estimate of acked bitrate.
2016-10-27 15:19:34 +00:00
video
Revert of Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #4 id:60001 of https://codereview.webrtc.org/2455063002/ )
2016-10-28 11:48:53 +00:00
voice_engine
Revert of New statistics interface for APM (patchset #11 id:200001 of https://codereview.webrtc.org/2433153003/ )
2016-10-28 10:29:37 +00:00
.gitignore
…
BUILD.gn
Remove remnants of libsrtp1
2016-10-25 16:47:31 +00:00
call.h
Integrate FlexfecReceiveStream with Call.
2016-10-24 06:37:24 +00:00
codereview.settings
…
common_types.cc
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common_types.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +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
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
…
transport.h
…
typedefs.h
…
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_frame.h
Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 07:44:17 +00:00
video_receive_stream.h
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +00:00
video_send_stream.h
Implement framesEncoded stat in video send ssrc stats.
2016-10-24 09:09:39 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
…
webrtc.gyp
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +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.