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webrtc_m130/webrtc
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Zijie He 48e4d6d609 Add zijiehe@chromium.org as OWNERS in WebRTC DesktopCapturer related logic 
Bug: chromium:747738
Change-Id: Iff83e89862ee190d0442cb3463c1dea0b87eb4b4
Reviewed-on: https://chromium-review.googlesource.com/582028
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Sergey Ulanov <sergeyu@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19116}
2017-07-23 21:37:59 +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
Remove remains of webrtc/base
2017-07-19 17:40:47 +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
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2017-07-19 17:40:47 +00:00
logging
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
media
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
modules
Add zijiehe@chromium.org as OWNERS in WebRTC DesktopCapturer related logic
2017-07-23 21:37:59 +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
Introduce RtpTransportInternal and SrtpTransport.
2017-07-19 20:38:02 +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
Set target API to 18 for MediaCodecUtils.
2017-07-21 10:30:02 +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
Add parsing/serializing for MID RTP header extension.
2017-07-21 17:33:25 +00:00
video
Remove remains of webrtc/base
2017-07-19 17:40:47 +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
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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
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
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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
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PRESUBMIT.py
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README.chromium
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typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
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
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.