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peah b1c9d1de36 Avoid that previous settings in APM are overwritten by WebRtcVoiceEngine 
This CL ensures that any previously set nondefault settings in the
audio processing module are not overwritten by the ApplyOptions
method in WebRtcVoiceEngine

BUG=webrtc:8018

Review-Url: https://codereview.webrtc.org/2985633002
Cr-Commit-Position: refs/heads/master@{#19144}
2017-07-25 22:45:24 +00:00
..
api
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
audio
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
base
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
call
Get rid of unnecessary cast of FlexfecReceiveStreamImpl to FlexfecReceiveStream
2017-07-25 13:40:06 +00:00
common_audio
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
common_video
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
examples
iOS AppRTCMobile: Close peerconnection when disconnecting
2017-07-24 14:32:33 +00:00
logging
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
media
Avoid that previous settings in APM are overwritten by WebRtcVoiceEngine
2017-07-25 22:45:24 +00:00
modules
Add jamiewalch to OWNERS.
2017-07-25 21:37:07 +00:00
ortc
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
p2p
Revert of Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting. (patchset #2 id:20001 of https://codereview.webrtc.org/2983213002/ )
2017-07-21 19:59:46 +00:00
pc
When a track is added/removed directly to MediaStream notify observer->OnRenegotionNeeded
2017-07-25 00:00:25 +00:00
rtc_base
Revert of Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting. (patchset #2 id:20001 of https://codereview.webrtc.org/2983213002/ )
2017-07-21 19:59:46 +00:00
rtc_tools
Print general usage information for event_log_analyzer
2017-07-20 15:05:09 +00:00
sdk
[iOS] Fix incorrectly oriented frames when rapidly switching between cameras.
2017-07-25 16:35:55 +00:00
stats
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
system_wrappers
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
test
Rename isolated_output to test_output and add a method to get the test_output directory.
2017-07-25 14:31:18 +00:00
video
Report minimum PSNR in VideoQualityTest and save corresponding frame to file
2017-07-25 12:45:03 +00:00
voice_engine
Move total audio energy and duration tracking to AudioLevel and protect with existing critial section.
2017-07-20 16:57:42 +00:00
.gitignore
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BUILD.gn
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
codereview.settings
…
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Add parsing/serializing for MID RTP header extension.
2017-07-21 17:33:25 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
DEPS
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LICENSE
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LICENSE_THIRD_PARTY
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OWNERS
…
PATENTS
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PRESUBMIT.py
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README.chromium
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typedefs.h
…
video_receive_stream.h
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +00:00
webrtc.gni
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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.