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webrtc_m130/webrtc
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nisse aec49d2b49 Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ ) 
Reason for revert:
Broke Chrome build, see, e.g.,
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/16237

Original issue's description:
> Delete deprecated and transitional stuff related to video frame refactoring.
>
> BUG=webrtc:5880
>
> Review-Url: https://codereview.webrtc.org/2622263002
> Cr-Commit-Position: refs/heads/master@{#17928}
> Committed: 713a3bbcc7

TBR=mflodman@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5880

Review-Url: https://codereview.webrtc.org/2845333002
Cr-Commit-Position: refs/heads/master@{#17929}
2017-04-28 13:03:40 +00:00
..
api
Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ )
2017-04-28 13:03:40 +00:00
audio
Have AudioSendStream register CNG payload types with the RtpRtcpModule.
2017-04-27 15:03:42 +00:00
base
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
2017-04-28 12:24:50 +00:00
call
Injectable audio encoders: WebRtcVoiceEngine and company
2017-04-27 09:08:52 +00:00
common_audio
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
2017-04-28 12:24:50 +00:00
common_video
Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ )
2017-04-28 13:03:40 +00:00
examples
Update FlexFEC field trial string in AppRTCMobile Android.
2017-04-28 09:52:20 +00:00
logging
Creating webrtc:video_stream_api
2017-04-26 06:42:15 +00:00
media
Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ )
2017-04-28 13:03:40 +00:00
modules
Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ )
2017-04-28 13:03:40 +00:00
ortc
Enabling 'gn check' on webrtc/video.
2017-04-25 07:35:35 +00:00
p2p
Don't crash if STUN error message is missing ERROR-CODE attribute.
2017-04-26 16:21:22 +00:00
pc
Run some peer connection end-to-end tests with an empty audio decoder factory
2017-04-27 10:55:57 +00:00
sdk
Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ )
2017-04-28 13:03:40 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
2017-04-28 12:24:50 +00:00
test
Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ )
2017-04-28 13:03:40 +00:00
tools
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ )
2017-04-26 10:38:35 +00:00
video
Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ )
2017-04-28 13:03:40 +00:00
voice_engine
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
2017-04-28 12:24:50 +00:00
.gitignore
…
BUILD.gn
Creating webrtc:video_stream_api
2017-04-26 06:42:15 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
config.cc
Fix default extension id for video content type extension
2017-04-26 09:35:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +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_frame.h
Revert of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #17 id:320001 of https://codereview.webrtc.org/2622263002/ )
2017-04-28 13:03:40 +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 number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +00:00
webrtc.gni
Remove the expat and jsoncpp dependencies to reduce binary size.
2017-04-24 19:05:06 +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.