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solenberg 87abc289a9 Add kwiberg@webrtc.org as root owner. 
NOTRY=true

Review-Url: https://codereview.webrtc.org/2060923003
Cr-Commit-Position: refs/heads/master@{#13125}
2016-06-14 09:39:40 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
Delete left-over files.
2016-06-14 08:54:52 +00:00
audio
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
base
Delete left-over files.
2016-06-14 08:54:52 +00:00
build
GN: Add modules_unittests
2016-06-13 07:19:53 +00:00
call
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
common_audio
GN: Add common_audio_unittests and common_video_unittests
2016-06-08 08:28:15 +00:00
common_video
Refactor scaling.
2016-06-13 11:06:14 +00:00
examples
A missing path separator caused aecdump recordings
2016-06-14 08:34:55 +00:00
libjingle
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
media
Delete left-over files.
2016-06-14 08:54:52 +00:00
modules
Optimize the repeated calls to AudioEffect.queryEffects() on Android
2016-06-13 19:05:30 +00:00
p2p
Delete left-over files.
2016-06-14 08:54:52 +00:00
pc
Removing obsolete method from channel.h.
2016-06-13 17:49:16 +00:00
sdk
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
system_wrappers
GN: Fix errors in rtc_include_tests conditions
2016-06-06 21:04:16 +00:00
test
Revert of Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420) (patchset #2 id:20001 of https://codereview.webrtc.org/2061723002/ )
2016-06-13 21:11:09 +00:00
tools
Delete some unused header files.
2016-06-10 11:41:13 +00:00
video
Update RateStatistics to handle too-little-data case.
2016-06-10 20:13:33 +00:00
voice_engine
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
.gitignore
…
audio_receive_stream.h
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
audio_send_stream.h
…
audio_sink.h
…
audio_state.h
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BUILD.gn
GN: Add modules_unittests
2016-06-13 07:19:53 +00:00
call.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
codereview.settings
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common_types.cc
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
common_types.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
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config.cc
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
config.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
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LICENSE
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LICENSE_THIRD_PARTY
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OWNERS
Add kwiberg@webrtc.org as root owner.
2016-06-14 09:39:40 +00:00
PATENTS
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PRESUBMIT.py
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README.chromium
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rtc_unittests_apk.isolate
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rtc_unittests.isolate
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supplement.gypi
Roll chromium_revision 086802955f..7fa6701bc5 (396351:398458)
2016-06-08 11:05:27 +00:00
transport.h
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typedefs.h
Remove runtime NEON detection
2016-05-17 17:56:48 +00:00
video_decoder.h
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video_encoder.h
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video_engine_tests_apk.isolate
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video_engine_tests.isolate
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video_frame.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
video_receive_stream.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
video_send_stream.h
…
webrtc_examples.gyp
Remove metrics_default from rtc_media dependencies.
2016-06-01 12:28:57 +00:00
webrtc_nonparallel_tests_apk.isolate
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webrtc_nonparallel_tests.isolate
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webrtc_perf_tests_apk.isolate
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webrtc_perf_tests.isolate
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webrtc_tests.gypi
GN: Add SDK tests to rtc_unittests.
2016-06-08 08:14:22 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.