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peah 64d6ff77ff In VoiceEngine, the settings for APM are applied in such a way that 
the previously specified setting is changed if it is specified to be changed,
and otherwise the previously specified setting is kept as it is.

This CL replicates this functionality for the way that the new APM
parameter scheme is used.

BUG=webrtc::6220, webrtc::6296, webrtc::6297, webrtc::6181, webrtc::5298

Review-Url: https://codereview.webrtc.org/2489343002
Cr-Commit-Position: refs/heads/master@{#15167}
2016-11-21 14:28:23 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Correctly pass drawn frame size when layout aspect ratio is used in EglRenderer.
2016-11-21 10:24:53 +00:00
audio
Expose RtpCodecParameters to VoiceMediaInfo stats.
2016-11-18 07:43:39 +00:00
base
Remove unused files linux.cc/.h and linuxfdwalk.c/.h.
2016-11-19 10:00:52 +00:00
build
iOS: Cleanup buildbot JSON files + bump iOS version to 10.0
2016-11-21 08:41:35 +00:00
call
Passed AudioMixer to AudioState::Config.
2016-11-17 14:48:56 +00:00
common_audio
Revert of Move smoothing filter to common audio. (patchset #3 id:60001 of https://codereview.webrtc.org/2484153002/ )
2016-11-18 09:31:19 +00:00
common_video
Remove all references to GYP
2016-11-16 19:11:38 +00:00
examples
Add Datachannel support to Android AppRTCMobile
2016-11-18 08:11:04 +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
In VoiceEngine, the settings for APM are applied in such a way that
2016-11-21 14:28:23 +00:00
modules
Initial rate allocation should not use fps = 0
2016-11-21 13:42:04 +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
iOS: Add AudioSendSideBwe field trial.
2016-11-17 18:29:50 +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
Changed the way we find the ProjectRootPath.
2016-11-18 15:06:47 +00:00
tools
Remove all references to GYP
2016-11-16 19:11:38 +00:00
video
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
2016-11-18 09:34:14 +00:00
voice_engine
Support multiple timestamp rates for sending DTMF.
2016-11-17 13:25:45 +00:00
.gitignore
…
BUILD.gn
Remove unused files linux.cc/.h and linuxfdwalk.c/.h.
2016-11-19 10:00:52 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
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
Remove Absolute Send Time from list of supported header extensions for audio streams.
2016-11-17 14:26:59 +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
Remove all references to GYP
2016-11-16 19:11:38 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
…
typedefs.h
…
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
2016-11-18 09:34:14 +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.