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minyue-webrtc 2d54784d89 Reland "Adding ANA config event to debug dump." 
Originally review in https://chromium-review.googlesource.com/c/535554/

Reverted in https://chromium-review.googlesource.com/c/539737/ due to upstreaming failure.

BUG=webrtc:7854

Change-Id: Ie4ad6ecfaf0f6b556dc662512d0be8ce94f8a4a8
Reviewed-on: https://chromium-review.googlesource.com/541436
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18865}
2017-06-30 14:32:23 +00:00
..
api
Enable the injection of an APM into a peerconnection
2017-06-30 14:24:04 +00:00
audio
Always ResetSenderCongestionControlObjects before RegisterEtc...
2017-06-30 08:38:56 +00:00
base
Delete SignalThread class.
2017-06-29 13:21:20 +00:00
call
Remove RtpDemuxer tweak for preventing multiple RSID inspections
2017-06-30 13:26:54 +00:00
common_audio
Put attribute before function name instead of after, as required by GCC
2017-06-26 08:32:40 +00:00
common_video
Revert "VideoFrameBuffer: Remove deprecated functions"
2017-06-29 13:39:35 +00:00
examples
Improves audio-routing in combination with BT in AppRTCMobile on Android.
2017-06-30 13:25:25 +00:00
logging
Limit the number of simultaneous event logs.
2017-06-28 13:40:51 +00:00
media
Enable -Wunused-function warning everywhere.
2017-06-30 14:05:22 +00:00
modules
Reland "Adding ANA config event to debug dump."
2017-06-30 14:32:23 +00:00
ortc
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +00:00
p2p
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
pc
Enable the injection of an APM into a peerconnection
2017-06-30 14:24:04 +00:00
rtc_base
Revert "Update includes for webrtc/{base => rtc_base} rename (3/3)"
2017-06-30 13:14:47 +00:00
rtc_tools
Fix android video_quality_loopback_test
2017-06-30 12:12:09 +00:00
sdk
Properly export the symbols of video frame-buffer classes for link-time
2017-06-29 16:11:10 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
More gracefully handle rtp timestamp jumps in the rtp to ntp estimator.
2017-06-30 13:28:13 +00:00
test
Opus implementation of the AudioDecoderFactoryTemplate API
2017-06-30 12:24:56 +00:00
tools
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
video
Fix receiving FlexFEC in video_loopback.
2017-06-30 09:34:20 +00:00
voice_engine
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +00:00
.gitignore
…
BUILD.gn
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Allow parsing empty RTCP TargetBitrate messages, but stop sending them.
2017-06-22 12:40:25 +00:00
common_types.h
Allow parsing empty RTCP TargetBitrate messages, but stop sending them.
2017-06-22 12:40:25 +00:00
config_unittest.cc
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
config.cc
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
config.h
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
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_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Add field trial for balanced degradation preference.
2017-06-14 12:52:21 +00:00
webrtc.gni
Adds detection of audio glitches for playout on iOS (reland)
2017-06-13 16:00:18 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.