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peah 8a8ebd94b0 Field trial support to whenever possible turn off the AGC and HPF 
When operating on mobile devices, where hardware support is available
for the AEC and NS functionality, it is desirable to be able to
operate without hardcoded behaviors for the WebRTC AGC and HPF.

This CL adds support to allow a field trial to turn these off
whenever that is possible.

BUG=webrtc:6220, webrtc:6183, webrtc:6181

Review-Url: https://codereview.webrtc.org/2876133002
Cr-Commit-Position: refs/heads/master@{#18226}
2017-05-22 22:48:47 +00:00
..
api
Renaming probing_interval to bwe_period globally.
2017-05-18 21:32:41 +00:00
audio
Renaming probing_interval to bwe_period globally.
2017-05-18 21:32:41 +00:00
base
Fixing potential AsyncInvoker deadlock that occurs for "reentrant" invocations.
2017-05-22 22:32:51 +00:00
call
Replace AudioSendStream::Config with rtclog::StreamConfig.
2017-05-22 17:12:26 +00:00
common_audio
Moving compile_assert_c.h to webrtc/base
2017-05-15 15:06:47 +00:00
common_video
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
examples
AppRTCMobile support to turn off the WebRTC AGC and HPF
2017-05-18 22:09:33 +00:00
logging
Replace AudioSendStream::Config with rtclog::StreamConfig.
2017-05-22 17:12:26 +00:00
media
Field trial support to whenever possible turn off the AGC and HPF
2017-05-22 22:48:47 +00:00
modules
This CL introduces a new APM sub-module named AGC2 that does not use the band
2017-05-22 13:57:06 +00:00
ortc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
p2p
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
pc
Add media related stats (audio level etc.) to unsignaled streams.
2017-05-19 20:09:47 +00:00
sdk
Field trial support to whenever possible turn off the AGC and HPF
2017-05-22 22:48:47 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Adding target to track asm_defines.h
2017-05-15 21:59:17 +00:00
test
Renaming probing_interval to bwe_period globally.
2017-05-18 21:32:41 +00:00
tools
Replace AudioSendStream::Config with rtclog::StreamConfig.
2017-05-22 17:12:26 +00:00
video
Revert of Reuse allocated encoders in SimulcastEncoderAdapter. (patchset #15 id:320001 of https://codereview.webrtc.org/2830793005/ )
2017-05-19 17:33:03 +00:00
voice_engine
Replace AudioSendStream::Config with rtclog::StreamConfig.
2017-05-22 17:12:26 +00:00
.gitignore
…
BUILD.gn
Revert of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2890733003/ )
2017-05-17 13:52:58 +00:00
call.h
…
codereview.settings
…
common_types.cc
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Delete RawVideoType enum, use the VideoType enum instead.
2017-04-28 14:18:05 +00:00
config.cc
Fix default extension id for video content type extension
2017-04-26 09:35:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +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_frame.h
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Update adaptation stats to support degradations in both resolution and framerate.
2017-05-16 06:40:18 +00:00
webrtc.gni
Replace sanitizers:deps with exe_and_shlib_deps (webrtc)
2017-05-19 06:51:46 +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.