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0ba43b5a2045b0b6de4c8e374895462961330b85
platform-external-webrtc/webrtc
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sakal 0ba43b5a20 Add support for adding VideoSinks to VideoTracks. 
VideoSinks receive the new kind of VideoFrames and will replace
VideoRenderers. Converting from old texture frames to VideoFrames will
involve conversion to I420 so it is not recommended to use VideoSinks
before all sources produce VideoFrames.

BUG=webrtc:7749, webrtc:7760

Review-Url: https://codereview.webrtc.org/3002553002
Cr-Commit-Position: refs/heads/master@{#19335}
2017-08-14 12:17:49 +00:00
..
api
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +00:00
audio
Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
2017-07-26 14:28:51 +00:00
base
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
call
Removed an unused variable from CallPerfTest::TestAudioVideoSync()
2017-08-11 09:08:33 +00:00
common_audio
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
common_video
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +00:00
examples
Add --canary flag to gradle project test and update gradle.
2017-08-11 08:03:13 +00:00
logging
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
media
Removed unused async_invoker_ in WebRtcVideoCapturer
2017-08-14 09:51:17 +00:00
modules
Remove older AEC-dump interface.
2017-08-14 10:35:40 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
pc
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
rtc_base
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
rtc_tools
Use default header extension map in rtc_event_log2text
2017-08-08 12:40:26 +00:00
sdk
Add support for adding VideoSinks to VideoTracks.
2017-08-14 12:17:49 +00:00
stats
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
system_wrappers
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
test
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +00:00
video
Reland of quest keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.chromium.org/2994043002/ )
2017-08-11 10:41:44 +00:00
voice_engine
Add thread annotations to AudioLevel
2017-08-10 12:01:21 +00:00
.gitignore
…
BUILD.gn
Removing NO_SOUND_SYSTEM
2017-08-14 07:40:58 +00:00
codereview.settings
…
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
config.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
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
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +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
Removing unused declared arg
2017-08-03 17:45:31 +00:00
whitespace.txt
…

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.