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webrtc_m130/webrtc
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peah 8f9ce1d991 Corrected the limit on the allowed API jitter in AEC3 
This CL loosens the requirement on the API jitter in APM
that can be tolerated without affecting the AEC3 performance.

BUG=webrtc:7911,chromium:738323

Review-Url: https://codereview.webrtc.org/2967493004
Cr-Commit-Position: refs/heads/master@{#18849}
2017-06-30 10:13:21 +00:00
..
api
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +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
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +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
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
logging
Limit the number of simultaneous event logs.
2017-06-28 13:40:51 +00:00
media
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
modules
Corrected the limit on the allowed API jitter in AEC3
2017-06-30 10:13:21 +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
Fix -Wcomment warning in webrtcsdp.cc
2017-06-30 06:19:31 +00:00
rtc_base
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
rtc_tools
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +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
Prevent data race in GetStaticInstance
2017-05-29 07:01:14 +00:00
test
Allow an external audio processing module to be used in WebRTC
2017-06-29 15:32:09 +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
…
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.